Hi everybody
has anybody succeded in vzaphfc installation? I need more info about it.
Thank you
Regards
Mauro
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Hi
I'm running asterisk 1.0.10 version. I've got 5 Linksys SPA 922 ip
phones. My problem is setting up my voice mail. I cant access it.
Please help
Regards
Khayelihle
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Hi Lito,
It depends on how you asked your telco provider to configure your 3 direct
lines. We called it trunking the 3 lines with pilot number. Telco can figure
it the way we configure our Asterisk Followme. (seized all, random, sequencial).
Regards,
Angel
Lito Lampitoc [EMAIL PROTECTED]
On Thu, Mar 29, 2007 at 09:04:27AM +0200, Khayelihle Mbona wrote:
Hi
I'm running asterisk 1.0.10 version. I've got 5 Linksys SPA 922 ip
phones. My problem is setting up my voice mail. I cant access it.
Please help
On which storage, exactly?
--
Tzafrir Cohen
Hi Ronald,
I can make my gtalk client connect with my Asterisk 1.4.1 infact jabber
show connected CLI command shows one user connected.
When I call I get no voice: I suspect it is from gtalk server to my
Asterisk because if I send a voice mail via gtalk I hear the female
voice who tells me to
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jordan
Novak
Okay, I get it. I still have a problem though. I have no way to wire
30%
of these end-points. P{hysically impossible. They do have cat3 twisted
pair to
each phone. But of course they want IP.
What phones do 'they'
I've got a problem with a SIP Account I am trying to dial in with. The
correct extension rings but when I pick up the call is not made and I
get a busy signal. Dialing out works just fine - just calling this
number doesn't seem to work.
Any pointers?
Thx
Michael
excerpt from sip.conf:
Hi Carlos,
type: *ps -A -F | grep panel*
You should see something like:
root 14851 1 0 2700 8164 0 11:01 ?00:00:01
/usr/bin/perl -w /usr/local/op_panel/op_server.pl -d -p
/var/run/asterisk/op_panel.pid
This means that tha panel process is running.
Giorgio Incantalupo
Hi Michael,
have you tried to set canreinvite = no inside incoming calls context in
sip.conf? Some SIP provider does not like reinvite.
Giorgio Incantalupo
Michael Zoller wrote:
I've got a problem with a SIP Account I am trying to dial in with. The
correct extension rings but when I pick
Hi Am,
I've got a similar problemyou mean you can connect and call but hear
no sound??
Giorgio
Am Turnip wrote:
I am trying to dial a GTalk, ie @gmail.com, address. I inscribed this
address in jabber.conf on the buddy= line. Upon executing the Dial application, I hear
only a brief
Giorgio Incantalupo wrote:
have you tried to set canreinvite = no inside incoming calls context
in sip.conf? Some SIP provider does not like reinvite.
That seemed to have done the trick - Thank you!
Michael
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Olivier wrote:
No, I'm far from inventing features, yet ! ;-)
It's a feature offered by Alcatel and I wanted to find in documentation,
a way to reproduce it, just in case I'm asked to do so.
I think it's the equivalent of call screening, but from caller perspective.
Cheers
Well I don't like
The tomshardware-guys (no gals would do this...) have removed the
fans, and immersed the innards of the computer in a sealed cabinet
filled with cooking oil. So they have a completely silent machine
in 40C warm oil. Amazing...
It certainly is. And, I suppose, this will work, for a
OPTIONS/200 messages looks correct. Yes, Asterisk requires the From:
header field to contain a valid extension to respond with a 200 to a OPTIONS
request (else it'll respond with a 404).
Raj
On 3/28/07, Steve Edwards [EMAIL PROTECTED] wrote:
I'm (still) trying to get my Asterisk box talking
Hi,
This is really strange (but probably simple solution).
The CALLERID(all) setting doesn't seem to work when the incomming
callerid is 'unknown'.
Dialplan looks like this:
exten = _3072,1,Answer
exten = _3072,n,Set(CALLERID(all)=DIRECT 0850553072)
exten =
Lee Jenkins wrote:
You have to login into the AMI server with proper credentials and send
commands.
Hi Lee!
Thank you for your mail. I do login to the Asterisk 5038 port and I have
all credentials. What I'm asking is what Action allows me to execute
system command.
What I tried is this:
PB == Peter Bowyer [EMAIL PROTECTED] writes:
PB No it can't - the latest registration 'wins'. To achieve
PB simutaneous ringing of more than one phone (hard or soft), you
PB need a SIP account for each and an entry in the dialplan which
PB rings both.
Indeed, this is the limitation of Asterisk
Ive installed asterisk and freepbx. Through the interface ive
configured 2 extensions, 6000 and 6001.
My problem is that when i try to call from extension 6000 to 6001, i
hear this msg Im-sorryan-error-has-occured and the call is
terminated.
As expected if i call to another number i get an error.
Hi Giorgio.
when i type: ps -A -F | grep panel:
#
[EMAIL PROTECTED]:/home/hernandezz# ps -A -F | grep panel
1000 5378 1 0 8596 15480 0 11:26 ?00:00:02
gnome-panel --sm-client-id default1
1000 5433 1 0 5903 10900 0 11:26 ?00:00:00
On Thu, 29 Mar 2007, Carlos Jerónimo wrote:
Ive installed asterisk and freepbx. Through the interface ive
configured 2 extensions, 6000 and 6001.
My problem is that when i try to call from extension 6000 to 6001, i
hear this msg Im-sorryan-error-has-occured and the call is
terminated.
As
Hey there folks,
Looking to my favorite mailing list for assistance and have a few bucks to
pay you for your time.
Me: Played with asterisk for a while in the early days and getting stuck on
silly stuff on a time sensitive project for a friend.
Project:
PSTN incoming call to asterisk and then
How to configure cisco 7902 with asterisk ,if you please can send me step by
step configuration steps .
Thanks
*
No employee or agent is authorized to conclude any binding agreement on behalf
of Xplorium with another party by e-mail without
What output does the CLI generate when you try to make a call? It will tell
you what the system is doing, so it will usually give you a good indicator
of what is causing the call to fail.
Alex
On 3/29/07, Carlos Jerónimo [EMAIL PROTECTED] wrote:
Ive installed asterisk and freepbx. Through the
Hi Carlos,
if you have not op_panel.pid in /var/run/asterisk this means the panel
server is not working.
I do not know where freepbx puts oppanel files (usually they are in
/usr/local but not always). Just find them and exec the file
*op_server.pl* in stand alone mode (just type ./op_server.pl
The zttest program results in 99%.
Jim
Tzafrir Cohen [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
On Tue, Mar 27, 2007 at 09:30:36PM -0400, Jim Duda wrote:
Lacy,
I don't have any zaptel cards installed. I do however have ztdummy
installed.
Is there some tweaks to ztdummy
Hi,
I have created mail to fax with txfax. I send faxes successfully. But I
want to know, is it possible to get result from txfax and what are solutions
to do this? For example, if fax send is unsuccessfully, the user will get
email, that fax was unsuccessfully.
Thanks
Am Donnerstag, den 29.03.2007, 15:04 +0300 schrieb Khaled Chehab:
How to configure cisco 7902 with asterisk ,if you please can send me
step by step configuration steps .
Khaled,
you already have a 7905 and a 7960, your older posts suggest that. Try
to configure the 7902 the same way. If
Hey everyone,
we are using several TE410 cards in a production environment that are
connected to several operators PRI's and it works great..
For one of the operators we have seen some strange problems in cdr
mismatches however.
Our cdr's show phonecalls that are disconnected at a certain
Just to let you know that this doesn't work with the latest SNOM
firmware.
We use 6.2.3 which works fine.
-- Jamie
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carsten
Bock
Sent: 27 March 2007 12:19
To: Asterisk Users Mailing List - Non-Commercial
Just be careful with the sidecar. It was to be screwed on and the screws
that come with the unit strip very easily. Make sure you have a nice
electronics grade screwdriver with a long thin shaft or you'll have
trouble with the side car.
Another really nice feature of this phone is that the BLF
Hi Murphy,
Sorry! But I didnt understand you :(
Can you give me an example?
When I talked about creating a new property in the events to return the
ActionID command, I just give an idea. My problem is to identify WHO has
raised the event.
Thanks!
Moacir O. de Souza Junior
Belo Horizonte -
bounjour
je dispose de differents commutateurs de LG (IP LDK) sur differents sites.
je voudrais savoir comment je pourrais interconnecter ces differents IP LDK
a un serveur Asterisk via IP ( ceci sous entend que chacun de ces
commutateurs dispose déjà d'une carte VOIBE).
Mecri d'avance pour
13/15 is good, and Im not saying this is definately the issue, but it's
just a thought I had. Download pingplotter (pingplotter.com) and run it to
the teliax server for about 30 minutes, or so. Set your test interval to
1second.See if it makes a nice steady line, or if it looks like a city
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Jan,
Is this call from PSTN? Probably the Nr is prohibited in PSTN, then
asterisk doesn't set the CALLERID. Try this:
exten = _3072,1,Answer
exten = _3072,n,SetCallerPres(allowed)
exten = _3072,n,Set(CALLERID(all)=DIRECT 0850553072)
Look here:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I'm using the biult in feature attended transfer. If someone calls me, I
hit the #, dial another extension and connect these two extensions. When
hitting # and dialing the nr, asterisk only diales the new nr for 15
seconds. Is it possible to
Most people download Asterisk, buy a bunch of phones and then run into
a brick wall, he said. Those 'Asterisk rescues' are a lot of our business
right now.
Most people? This guy's got a marketing dude's hand up his shirt. Do
you know many businesses that installed their own Norstar or
On Thu, Mar 29, 2007 at 08:28:53AM -0400, Jim Duda wrote:
The zttest program results in 99%.
So you have a working timing source. No need to waste your time here.
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406
On Thu, 2007-03-29 at 13:26 +0200, Remco Barendse wrote:
On Thu, 29 Mar 2007, Carlos Jerónimo wrote:
Ive installed asterisk and freepbx. Through the interface ive
configured 2 extensions, 6000 and 6001.
My problem is that when i try to call from extension 6000 to 6001, i
hear this msg
Hi Chris,
Yes the call was from PSTN and your solution worked great! I've read about
SetCallerPres earlier but I didn't connect the dots this time.
Thanks alot! :)
Regards,
Jan
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Christoph Fürstaller
Both end devices should be using same codecs. set dtmf = rfc2833 and set
canreinvite = no in sip.conf for both endpoints. This should solve the
problem. you should also check which codecs support rfc2833 for dtmf and
use that codec.
On 3/29/07, Gordon Henderson [EMAIL PROTECTED] wrote:
On
I need a softphone - for usb phone devices - that I can alter (insert logo,
menu, etc).
Does somebody know such one?
[]s
--
Abraços
Luis Claudio
Mobile + 55 21 9215 2888
Mobile +55 15 9141 8402
Office +55 15 2102 5859
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Hi,
Maybe this question has already been answered but I could find its answer
anywhere.
From here http://www.soft-switch.org/downloads/spandsp/, I can see a new
0.0.3pre28 version of spandsp have been added in march.
When you open this archive file, you can see a ChangeLog file but its
Hello Asterisk users,
Can someone thwack me with a clue stick please?
I am following the Asterisk TFOT book Dial() example trying to get the limit
and announcements to work as per below.
These settings seem to have no effect.
There are no warning messages after 4 minutes or every 30 secs
Hi,
what could be the maximum simultaneous calls can asterisk do? i read about
the asterisk business edition review[1] and it can only handle 120
simultaneous calls? i'm using 1.2.x branch of asterisk and i use more or
less 90 simultaneous calls.
[1]
Mark Reardon wrote:
Hello Asterisk users,
Can someone thwack me with a clue stick please?
I am following the Asterisk TFOT book Dial() example trying to get the
limit
and announcements to work as per below.
These settings seem to have no effect.
There are no warning messages after 4 minutes
Hi,
I'm testing sip trunking on Asterisk (v1.4.0-beta3) with various voip
service providers and stumbled on this
issue. This very well may be a known issue or something misconfigured in
my extensions.conf/sip.conf files.
The service provider requires registration and authentication. The
On Thu, 2007-03-29 at 15:51 +0100, Mark Reardon wrote:
Hello Asterisk users,
Can someone thwack me with a clue stick please?
I am following the Asterisk TFOT book Dial() example trying to get the
limit and announcements to work as per below.
These settings seem to have no effect.
Hi,
I know this question doesn't exactly relate to the core of this list but I
thought it does relate to its hacker spirit.
Is it possible to install a Cisco Call Manager 5.X on a non-Cisco appliance
?
A friend of mine working for a Cisco VAR told me his colleagues couldn't
make it, even for
Ahhh - in the tfot book they use [] not ().
Thanks a million
On 3/29/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Mark Reardon wrote:
Hello Asterisk users,
Can someone thwack me with a clue stick please?
I am following the Asterisk TFOT book Dial() example trying to get the
limit
Hi all
We run an * 1.2.4 under FreeBSD with ztdummy kernel module.
zttest reports 99.9something % of accuracy, so timing should be fine.
SIP connections work fine, but we have a strange problem with IAX2
connections.
When an IAX2 call originates from the FreeBSD Asterisk to another Asterisk,
- Chris Nighswonger [EMAIL PROTECTED] wrote:
That is the conclusion I came to and was confirmed today in a very
brief chat with one of the individuals listed as a developer on the
chan_skinny module. He said that they could be implemented.
What I would like to know, and do not
I hate SIP. The only reason I'm doing this is that its cheaper than
deploying the server to a colo facility. My provider has given me a
non-standard IP block, so I can't do typical routing.
I have Asterisk server - MT w\NAT - PPPoE - MT - Provider.
I setup a dst-nat on 5060 to the
Mark Quitoriano wrote:
what could be the maximum simultaneous calls can asterisk do? i read
about the asterisk business edition review[1] and it can only handle
120 simultaneous calls? i'm using 1.2.x branch of asterisk and i use
more or less 90 simultaneous calls.
Mark,
We are regularly
What do you mean by 'non-standard' IP block?
Is the Asterisk machine behind a NAT, or are only your clients?
Did you look at the nat setting sin sip.conf?
Do you have a static public address that can be routed to the Asterisk
box?
From: [EMAIL
Thanks Andrew, I understand the issue now.
Removing insecure=very allows the Grandstream phones to work, they
register separate lines on separate ports (eg Line 1=5060, Line 2=5062,
etc).
Unfortunately I cannot find a port setting for the Aastra 480i, I shall
get on their case.
regards,
Hello all,
I've got myself into a bizzare situation that I can't seem to get myself
out of... Was wondering if anyone had some advice that might get me 'over the
hill' on this...
Some background: PBX consists of an Asterisk box (running TrixBox), 4 Cisco
7960's, 2 Polycom IP500's, and
Tomislav Parcina wrote:
Lee Jenkins wrote:
You have to login into the AMI server with proper credentials and
send commands.
*snipped
OK, maybe he doesn't show output, so I have tried this:
Action: Command
Command: ! rm /tmp/test.txt
Response: Follows
Privilege: Command
--END COMMAND--
But
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Drew Gibson
Sent: 29 March 2007 17:13
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Multi-line phones - Asterisk
uses wrong callerid
Thanks Andrew, I
Digium
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Moacir O. de Souza Junior - Personalsoft Sistemas Ltda. wrote:
Hi Murphy,
Sorry! But I didn’t understand you :(
Can you give me an example?
When I talked about creating a new property in the events to return the
ActionID command, I just give an idea. My problem is to identify WHO has
raised
I have a block of 20 IP addresses, so I can't really carve out /30s and
whatnot to route between locations.
Asterisk is the client. I am doing Interop testing with some vendors before
I ship it out to a colo facility.
I have used the NAT setting with Asterisk as the server on the open
Developer
Digium
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Hi folks!
Does anybody know how to receive send faxes throw mISDN? It's almost
impossible!
Thanks
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Steve Langstaff wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Drew Gibson
Sent: 29 March 2007 17:13
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Multi-line phones - Asterisk
uses wrong callerid
I have the following scenario:
PSTN gateway (202.180.nnn.nnn) - OpenSER 1.0.1 (147.202.nnn.nnn) - Asterisk
1.2.16
(203.89.nnn.nnn)
When an incoming call is received, often (but not always) Asterisk repeatedly
sends a SIP 200 OK message and eventually hangs up the call.
sip.conf
[general]
port
(SIP/6009-08197f70, record-enable|6000|IN)
in new stack
-- Executing GotoIf(SIP/6009-08197f70, 0?2:4) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI(SIP/6009-08197f70,
recordingcheck|20070329-181220|1175188340.3) in new stack
-- Launched AGI Script /var/lib/asterisk/agi
hi every body,
tomorrow i remarked that when i call someone and after about 1min30s
or 2min, i heared a scrach noise for 10s to 15s and then i heared echo
any edia??
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Hey All,
This is more a polycom question, but involves how to make it
specifically interact with asterisk...
I have setup a special extension in asterisk which toggles whether our
Night Service is turned on. However, I would like to have 2 things
happen on our Polycom 501 (or 601)s. Where there
Richard,
When you said there were means that there is no way to do it anymore.
Thanks,
[]s
Moacir O. de Souza Junior
Belo Horizonte Minas Gerais - Brasil
-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de Richard Lyman
Enviada em: quinta-feira, 29 de
On Thu, 2007-03-29 at 12:16 -0400, Brad Stockdale wrote:
Hello all,
I've got myself into a bizzare situation that I can't seem to get myself
out of... Was wondering if anyone had some advice that might get me 'over the
hill' on this...
Some background: PBX consists of an Asterisk
how do these phones perform? ok for office use? work well with asterisk?
any info would be appreciated.
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--
Message: 3
Date: Thu, 29 Mar 2007 16:04:43 +0200
From: [EMAIL PROTECTED]
Subject: SV: [asterisk-users] Set(CALLERID(all) not working
Indeed this might be the failing point... Unfortunately, because I have no
Cisco CCO account anymore, I have no access to firmware... I will try to find
a copy of an old firmware for these phones. If I can find one, I hope it
fixes my problem.
Thanks,
Brad
Apologies in advance if this is a
Hi Steve, your sugestion is correct, but i registed 2 times in FreePbx
foruns this week, and my login is inactive yet. In the mail i receive
this msg:
Welcome to FreePBX Forums Forums
Please keep this email for your records. Your account information is as follows:
Your account is
Jerric wrote:
Please remove this email from your mailing list.
UNSUBSCRIBE
---cut---
List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,
mailto:[EMAIL PROTECTED]
---cut---
---cut---
To UNSUBSCRIBE or update options visit:
LKS GMAIL wrote:
Does anybody know how to receive send faxes throw mISDN? It's almost
impossible!
I know that IAXmodem users are doing it. They have to get the right
version of the mISDN stuff, though, I think.
Lee.
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This is my problem. I don't even know it this is the right site to ask
for it, but let's go.
I've a Asterisk box with 1.4.1 version version installed. It's equipped
with a TDM400 with 2 FXO modules and a HFC based ISDN BRI card.
Everything goes OK, except for the following scenario:
on the PMP
At 18:23 3/28/2007, Mike Diehl wrote:
Actually, it turns out that sometimes I can't get ANY DTMF to work. I can
call a local phone number and log into my voicemail system at work. But my
wife is unable to dial a toll free number and use their IVR. Hope this
helps.
What does your log read?
Hi Giorgi thanks for all, it works. Your opinion is correct, my
op_server was not run, and i run him.
I'm using Ubuntu Dapper and i want run the op_server when the machine
starts, and i add a line in the file rc.local like this:
cd /var/www/html/panel/; su -c /var/www/html/panel/op_server.pl
Thursday, March 29, 2007, 7:18:43 PM, LKS wrote:
Hi folks!
Does anybody know how to receive send faxes throw mISDN? It's almost
impossible!
Describe your problem, but read this before:
http://www.catb.org/~esr/faqs/smart-questions.html
It works for me, in 3 places, the analogue fax machines
Just a FYI to the list.
It seems that although BT only present 6 digits (as standard) for CLI
they expect the full number minus the leading 0 to set CLI.
So if a number is 01234 987654
They will present 987654
and you need to present to them 1234 987654
Hmmm
Steve
--
NetTek Ltd UK mob
TP'n to follow flow.
'there were' means that *over time*
enough mods added to be able to track most of the 'call flow' by it.
referring to the callerid name manipulation method
Moacir O. de Souza Junior - Personalsoft Sistemas Ltda. wrote:
Richard,
When you said there were means that there
One potential reason could be that the ACK request being sent to
Asterisk is malformed. Notice branch=0 in the top Via. This should start
with z9hG4bK magic cookie since the INVITE was an RFC 3261 transaction.
While branch=0 is valid in RFC 2543, I don't think an INVITE can start-off
as RFC 3261
I have the following scenario:
PSTN gateway (202.180.nnn.nnn) - OpenSER 1.0.1 (147.202.nnn.nnn) - Asterisk
1.2.16
(203.89.nnn.nnn)
When an incoming call is received, often (but not always) Asterisk repeatedly
sends a SIP 200 OK message and eventually hangs up the call.
sip.conf
[general]
port =
Mike Hammett wrote:
I hate SIP. The only reason I'm doing this is that its cheaper than
deploying the server to a colo facility. My provider has given me a
non-standard IP block, so I can't do typical routing.
I have Asterisk server - MT w\NAT - PPPoE - MT - Provider.
I setup a
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Brad Stockdale wrote:
Hello all,
I've got myself into a bizzare situation that I can't seem to get myself
out of... Was wondering if anyone had some advice that might get me 'over the
hill' on this...
Some background: PBX consists of
Hi,
we are searching for somebody who can translate German to English
Asterisk documentation. We have some generic Asterisk documentation
(e.g.
http://www.das-asterisk-buch.de ) which we would like to share with
the english speaking world. And it takes someone who understands
Asterisk
- Derek Whitten [EMAIL PROTECTED] wrote:
if i remember right, most of the buttons on those and the 12SP+ phones
don't really work
because there isn't a button template in *
There is a button template, the problem is that most of the softkeys simply
aren't implemented.
--
Jason Parker
Tomislav Parcina wrote:
Lee Jenkins wrote:
You have to login into the AMI server with proper credentials and send
commands.
Hi Lee!
Thank you for your mail. I do login to the Asterisk 5038 port and I have
all credentials. What I'm asking is what Action allows me to execute
system command.
On 3/29/07, Brad Stockdale [EMAIL PROTECTED] wrote:
Hello all,
loadInformation7 model=IP Phone 7960P003-08-6-00/loadInformation7
Should be POS03-08-6-00. The same as your .loads file. Also change
this in the OS79XX file.
P003-08-6-00.bin
P003-08-6-00.sbn
P0S3-08-6-00.loads
Hello mailing list,
I have been porting one of my Asterisk boxes to 1.4 and I have
encountered a nasty DTMF problem. What happens is someone might come
in to my IVR and enter 12345 and what will actually come through
could be along the lines of 12234445. Sometimes it works, sometimes
it
Situation, simple home setup:
* Trixbox 2.0
* Feature Codes installed
* GNet PA-168V based ATA
* Cheesy cordless analogue phone
From what I gather, dialing *70 from the handset should activate Call
Waiting. All it seems to do is change the message The person at
extension is on the phone to
I believe that's Roger Workman's job... I'll go kick him and see that he
activates you.
Alex
On 3/29/07, Carlos Jerónimo [EMAIL PROTECTED] wrote:
Hi Steve, your sugestion is correct, but i registed 2 times in FreePbx
foruns this week, and my login is inactive yet. In the mail i receive
this
Set(SIP/6009-08197f70, RT=15) in new stack
-- Executing Macro(SIP/6009-08197f70, record-enable|6000|IN)
in new stack
-- Executing GotoIf(SIP/6009-08197f70, 0?2:4) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI(SIP/6009-08197f70,
recordingcheck|20070329-181220
On Thu, 29 Mar 2007, Luis Claudio Santos wrote:
I need a softphone - for usb phone devices - that I can alter (insert logo,
menu, etc).
Does somebody know such one?
Maybe not quite what you want, but I used a Yealink USB phone with Linux
and there was a driver for it that would let you read
i´m trying find in the codes of asterisk as change the name of file created
after that a extension dial for a queue.
Has someone some sugestion for obtain this name of queue(150)?
Use Queue monitor rather than Agent monitor.
agents.conf
recordagentcalls=no
queues.conf
monitor-type =
I have a 501 with traditional power and a 301 with PoE. I rightfully assumed
that the traditional power from the 501 would work on the 301.
How do I get the PoE to work? Do I use the Polycom PoE cable in addition to
whatever PoE injection method I use? I have a Cisco PoE injector that works
on my
We have the caller id with name option enabled with our provider,
however, our polycom 501 phones will only display the number of the
incoming call. Is there a way to see the callerid name from the cli when
the call is coming in (like a print in the dial plan)? I'm not sure if
the problem is with
Try adding this to the [options] section of
/etc/asterisk/asterisk.conf:
internal_timing = yes
The restart asterisk. Let us know if it helps.
That solves the problem. Thanks!
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How do I set rxgain per channel on zapata.conf? I've a TDM400 with 2 FXS.
Thank you!
Santiago del Castillo
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